flycast/core/oslib/audiobackend_alsa.cpp

260 lines
6.0 KiB
C++

#include "oslib/audiostream.h"
#if USE_ALSA
#include <alsa/asoundlib.h>
#include "cfg/cfg.h"
static snd_pcm_t *handle;
static bool pcm_blocking = true;
static snd_pcm_uframes_t buffer_size;
static snd_pcm_uframes_t period_size;
// We're making these functions static - there's no need to pollute the global namespace
static void alsa_init()
{
snd_pcm_hw_params_t *params;
unsigned int val;
int dir=-1;
string device = cfgLoadStr("alsa", "device", "");
int rc = -1;
if (device == "" || device == "auto")
{
INFO_LOG(AUDIO, "ALSA: trying to determine audio device");
// trying default device
device = "default";
rc = snd_pcm_open(&handle, device.c_str(), SND_PCM_STREAM_PLAYBACK, 0);
// "default" didn't work, try first device
if (rc < 0)
{
device = "plughw:0,0,0";
rc = snd_pcm_open(&handle, device.c_str(), SND_PCM_STREAM_PLAYBACK, 0);
if (rc < 0)
{
device = "plughw:0,0";
rc = snd_pcm_open(&handle, device.c_str(), SND_PCM_STREAM_PLAYBACK, 0);
}
}
// first didn't work, try second
if (rc < 0)
{
device = "plughw:1,0";
rc = snd_pcm_open(&handle, device.c_str(), SND_PCM_STREAM_PLAYBACK, 0);
}
// try pulse audio backend
if (rc < 0)
{
device = "pulse";
rc = snd_pcm_open(&handle, device.c_str(), SND_PCM_STREAM_PLAYBACK, 0);
}
if (rc < 0)
INFO_LOG(AUDIO, "ALSA: unable to automatically determine audio device.");
}
else {
rc = snd_pcm_open(&handle, device.c_str(), SND_PCM_STREAM_PLAYBACK, 0);
}
if (rc < 0)
{
WARN_LOG(AUDIO, "ALSA: unable to open PCM device %s: %s", device.c_str(), snd_strerror(rc));
return;
}
INFO_LOG(AUDIO, "ALSA: Successfully initialized \"%s\"", device.c_str());
/* Allocate a hardware parameters object. */
snd_pcm_hw_params_alloca(&params);
/* Fill it in with default values. */
rc=snd_pcm_hw_params_any(handle, params);
if (rc < 0)
{
WARN_LOG(AUDIO, "ALSA: Error:snd_pcm_hw_params_any %s", snd_strerror(rc));
return;
}
/* Set the desired hardware parameters. */
/* Interleaved mode */
rc=snd_pcm_hw_params_set_access(handle, params, SND_PCM_ACCESS_RW_INTERLEAVED);
if (rc < 0)
{
WARN_LOG(AUDIO, "ALSA: Error:snd_pcm_hw_params_set_access %s", snd_strerror(rc));
return;
}
/* Signed 16-bit little-endian format */
rc=snd_pcm_hw_params_set_format(handle, params, SND_PCM_FORMAT_S16_LE);
if (rc < 0)
{
WARN_LOG(AUDIO, "ALSA: Error:snd_pcm_hw_params_set_format %s", snd_strerror(rc));
return;
}
/* Two channels (stereo) */
rc=snd_pcm_hw_params_set_channels(handle, params, 2);
if (rc < 0)
{
WARN_LOG(AUDIO, "ALSA: Error:snd_pcm_hw_params_set_channels %s", snd_strerror(rc));
return;
}
/* 44100 bits/second sampling rate (CD quality) */
val = 44100;
rc=snd_pcm_hw_params_set_rate_near(handle, params, &val, &dir);
if (rc < 0)
{
WARN_LOG(AUDIO, "ALSA: Error:snd_pcm_hw_params_set_rate_near %s", snd_strerror(rc));
return;
}
/* Set period size to settings.aica.BufferSize frames. */
period_size = settings.aica.BufferSize;
rc=snd_pcm_hw_params_set_period_size_near(handle, params, &period_size, &dir);
if (rc < 0)
{
WARN_LOG(AUDIO, "ALSA: Error:snd_pcm_hw_params_set_buffer_size_near %s", snd_strerror(rc));
return;
}
else
{
INFO_LOG(AUDIO, "ALSA: period size set to %ld", period_size);
}
buffer_size = (44100 * 100 /* settings.omx.Audio_Latency */ / 1000 / period_size + 1) * period_size;
rc=snd_pcm_hw_params_set_buffer_size_near(handle, params, &buffer_size);
if (rc < 0)
{
WARN_LOG(AUDIO, "ALSA: Error:snd_pcm_hw_params_set_buffer_size_near %s", snd_strerror(rc));
return;
}
else
{
INFO_LOG(AUDIO, "ALSA: buffer size set to %ld", buffer_size);
}
/* Write the parameters to the driver */
rc = snd_pcm_hw_params(handle, params);
if (rc < 0)
{
WARN_LOG(AUDIO, "ALSA: Unable to set hw parameters: %s", snd_strerror(rc));
return;
}
}
static u32 alsa_push(void* frame, u32 samples, bool wait)
{
if (wait != pcm_blocking) {
snd_pcm_nonblock(handle, wait ? 0 : 1);
pcm_blocking = wait;
}
int rc = snd_pcm_writei(handle, frame, samples);
if (rc == -EPIPE)
{
/* EPIPE means underrun */
snd_pcm_prepare(handle);
// Write some silence then our samples
const size_t silence_size = period_size * 4;
void *silence = alloca(silence_size * 4);
memset(silence, 0, silence_size * 4);
snd_pcm_writei(handle, silence, silence_size);
snd_pcm_writei(handle, frame, samples);
}
return 1;
}
static void alsa_term()
{
snd_pcm_drain(handle);
snd_pcm_close(handle);
}
std::vector<std::string> alsa_get_devicelist()
{
std::vector<std::string> result;
char **hints;
int err = snd_device_name_hint(-1, "pcm", (void***)&hints);
// Error initializing ALSA
if (err != 0)
return result;
// special value to automatically detect on initialization
result.push_back("auto");
char** n = hints;
while (*n != NULL)
{
// Get the type (NULL/Input/Output)
char *type = snd_device_name_get_hint(*n, "IOID");
char *name = snd_device_name_get_hint(*n, "NAME");
if (name != NULL)
{
// We only want output or special devices (like "default" or "pulse")
// TODO Only those with type == NULL?
if (type == NULL || strcmp(type, "Output") == 0)
{
// TODO Check if device works (however we need to hash the resulting list then)
/*snd_pcm_t *handle;
int rc = snd_pcm_open(&handle, name, SND_PCM_STREAM_PLAYBACK, 0);
if (rc == 0)
{
result.push_back(name);
snd_pcm_close(handle);
}
*/
result.push_back(name);
}
}
if (type != NULL)
free(type);
if (name != NULL)
free(name);
n++;
}
snd_device_name_free_hint((void**)hints);
return result;
}
static audio_option_t* alsa_audio_options(int* option_count)
{
*option_count = 1;
static audio_option_t result[1];
result[0].cfg_name = "device";
result[0].caption = "Device";
result[0].type = list;
result[0].list_callback = alsa_get_devicelist;
return result;
}
static audiobackend_t audiobackend_alsa = {
"alsa", // Slug
"Advanced Linux Sound Architecture", // Name
&alsa_init,
&alsa_push,
&alsa_term,
&alsa_audio_options
};
static bool alsa = RegisterAudioBackend(&audiobackend_alsa);
#endif